The purpose of this thesis is to examine the possibilities of creating a MPEG-4-compatible set-top box for IP-networks based on open standards. Existing alternatives for transporting MPEG-4 over IP are evaluated and ISMA is found to be an important actor within the area. ISMA is a non-profit corporation formed to provide a forum for the creation of specifications that define an interoperable implementation for streaming rich media over IP-networks. Two different designs based on ISMA's recommendation are constructed and evaluated. The designs have different levels of complexity and the more complex design is found to be better due to its extended functionality. During the design process a number of problems related to this kind of set-top box are discovered. It is believed, however, that many of these problems will be solved within the near future.

The imminent inception of third-generation (3G) mobile communication networks offers an unprecedented opportunity for the development of video streaming applications through wireless Internet access. Different design challenges exist in implementing video streaming connections spanning both wired and wireless domains. A split-domain TCP-friendly streaming video transmission protocol is presented based on adaptive rate encoding in the MPEG-4 video format. Network simulations are conducted to demonstrate the benefits and viability of such a video streaming scheme over existing options. Further feature enhancements and refinements are necessary for the proposed protocol to achieve its full potential.

In this thesis, a novel Scalable Multiple Description Coding (SMDC) framework is proposed. To address the bandwidth fluctuation, packet loss and heterogeneity problems in the wireless networks and further enhance the error resilience tools in Moving Pictures Experts Group 4 (MPEG-4), the joint design of layered coding (LC) and multiple description coding (MDC) is explored. It leverages a proposed distributed multimedia delivery mobile network (D-MDMN) to provide path diversity to combat streaming video outage due to handoff in Universal Mobile Telecommunications System (UMTS). The corresponding intra-RAN (Radio Access Network) handoff and inter-RAN handoff procedures in D-MDMN are studied in details, which employ the principle of video stream re-establishing to replace the principle of data forwarding in UMTS. Furthermore, a new IP (Internet Protocol) Differentiated Services (DiffServ) video marking algorithm is proposed to support the unequal error protection (UEP) of LC components of SMDC. Performance evaluation is carried through simulation using OPNET Modeler 9. 0. Simulation results show that the proposed handoff procedures in D-MDMN have better performance in terms of handoff latency, end-to-end delay and handoff scalability than that in UMTS. Performance evaluation of our proposed IP DiffServ video marking algorithm is also undertaken, which shows that it is more suitable for video streaming in IP mobile networks compared with the previously proposed DiffServ video marking algorithm (DVMA).

By equipping vehicles with the on-board wireless facility, the newly emerged vehicular networking targets to provision the broadband serves to vehicles. As such, a variety of novel and exciting applications can be provided to vehicular users to enhance their road safety and travel comfort, and finally raise a complete change to their on-road life. As the content distribution and media/video streaming, such as Youtube, Netflix, nowadays have become the most popular Internet applications, to enable the efficient content distribution and audio/video streaming services is thus of the paramount importance to the success of the vehicular networking. This, however, is fraught with fundamental challenges due to the distinguished natures of vehicular networking. On one hand, the vehicular communication is challenged by the spotty and volatile wireless connections caused by the high mobility of vehicles. This makes the download performance of connections very unstable and dramatically change over time, which directly threats to the on-top media applications. On the other hand, a vehicular network typically involves an extremely large-scale node population (e.g., hundreds or thousandths of vehicles in a region) with intense spatial and temporal variations across the network geometry at different times. This dictates any designs to be scalable and fully distributed which should not only be resilient to the network dynamics, but also provide the guaranteed quality-of-service (QoS) to users. The purpose of this dissertation is to address the challenges of the vehicular networking imposed by its intrinsic dynamic and large-scale natures, and build the efficient, scalable and, more importantly, practical systems to enable the cost-effective and QoS guaranteed content distribution and media streaming services to vehicular users. Note that to effective- ly deliver the content from the remote Internet to in-motion vehicles, it typically involves three parts as: 1.) an infrastructure grid of gateways which behave as the data depots or injection points of Internet contents and services to vehicles, 2.) protocol at gateways which schedules the bandwidth resource at gateways and coordinates the parallel transmissions to different vehicles, and 3.) the end-system control mechanism at receivers which adapts the receiver’s content download/playback strategy based on the available network throughput to provide users with the desired service experience. With above three parts in mind, the entire research work in this dissertation casts a systematic view to address each part in one topic with: 1.) design of large-scale cost-effective content distribution infrastructure, 2.) MAC (media access control) performance evaluation and channel time scheduling, and 3.) receiver adaptation and adaptive playout in dynamic download environment. In specific, in the first topic, we propose a practical solution to form a large-scale and cost-effective content distribution infrastructure in the city. We argue that a large-scale infrastructure with the dedicated resources, including storage, computing and communication capacity, is necessary for the vehicular network to become an alternative of 3G/4G cellular network as the dominating approach of ubiquitous content distribution and data services to vehicles. On addressing this issue, we propose a fully distributed scheme to form a large-scale infrastructure by the contributions of individual entities in the city, such as grocery stores, movie theaters, etc. That is to say, the installation and maintenance costs are shared by many individuals. In this topic, we explain the design rationale on how to motivate individuals to contribute, and specify the detailed design of the system, which is embodied with distributed protocols and performance evaluation. The second topic investigates on the MAC throughput performance of the vehicle-to- infrastructure (V2I) communications when vehicles drive through RSUs, namely drive-thru Internet. Note that with a large-scale population of fast-motion nodes contending the chan- nel for transmissions, the MAC performance determines the achievable nodal throughput and is crucial to the on-top applications. In this topic, using a simple yet accurate Marko- vian model, we first show the impacts of mobility (characterized by node velocity and moving directions) on the nodal and system throughput performance, respectively. Based on this analysis, we then propose three enhancement schemes to timely adjust the MAC parameters in tune with the vehicle mobility to achieve the maximal the system throughput. The last topic investigates on the end-system design to deliver the user desired media streaming services in the vehicular environment. In specific, the vehicular communications are notoriously known for the intermittent connectivity and dramatically varying throughput. Video streaming on top of vehicular networks therefore inevitably suffers from the severe network dynamics, resulting in the frequent jerkiness or even freezing video playback. To address this issue, an analytical model is first developed to unveil the impacts of network dynamics on the resultant video performance to users in terms of video start-up delay and smoothness of playback. Based on the analysis, the adaptive playout buffer mechanism is developed to adapt the video playback strategy at receivers towards the user-defined video quality. The proposals developed in the three topics are validated with the extensive and high fidelity simulations. We believe that our analysis developed in the dissertation can provide insightful lights on understanding the fundamental performance of the vehicular content distribution networks from the aspects of session-level download performance in urban vehicular networks (topic 1), MAC throughput performance (topic 2), and user perceived media quality (topic 3). The protocols developed in the three topics, respectively, offer practical and efficient solutions to build and optimize the vehicular content distribution networks.

The explosive growth of the Internet during the last few years, has been propelled by the TCP/IP protocol suite and the best effort packet forwarding service. However, quality of service (QoS) is far from being a reality especially for multimedia services like video streaming and video conferencing. In the case of wireless and mobile networks, the problem becomes even worse due to the physics of the medium, resulting into further deterioration of the system performance. Goal of this dissertation is the systematic development of comprehensive models that jointly characterize the performance of transport protocols and media delivery in heterogeneous wireless networks. At the core of our novel methodology, is the use of analytical models for driving the design of media transport algorithms, so that the delivery of conversational and non-interactive multimedia data is enhanced in terms of throughput, delay, and jitter. More speciffically, we develop analytical models that characterize the throughput and goodput of the transmission control protocol (TCP) and the transmission friendly rate control (TFRC) protocol, when CBR and VBR multimedia workloads are considered. Subsequently, we enhance the transport protocol models with new parameters that capture the playback buffer performance and the expected video distortion at the receiver. In this way a complete end-to-end model for media streaming is obtained. This model is used as a basis for a new algorithm for rate-distortion optimized mode selection in video streaming appli- cations. As a next step, we extend the developed models for the aforementioned protocols, so that heterogeneous wireless networks can be accommodated. Subsequently, new algorithms are proposed in order to enhance the developed media streaming algorithms when heterogeneous wireless networks are also included. Finally, the aforementioned models and algorithms are extended for the case of concurrent multipath media transport over several hybrid wired/wireless links.

For more than a decade, streaming media services, including on-line conferences, distance education and movie broadcasting, have gained much popularity on the Internet. Due to the high bandwidth requirements and long lived nature of video streaming, it requires huge transmission cost to support these streaming media services. In addition, how to adapt rich multimedia content to satisfy various resource-constrained devices presents a challenge. The limited and time-varying network bandwidth complicates the content adaptation tasks. Differentiated content delivery may be required to meet diverse client profiles and user preferences. Therefore, in order to reduce transmission cost to serve heterogeneous clients for efficient streaming, in this dissertation, several novel schemes including transcoding-enable proxy caching scheme, reactive transmission schemes, and network coding P2P content distribution scheme, are proposed to support efficient multiple-version and layered video delivery in the proxy-attached network environment as well as to provide efficient interactive IPTV service in a peer-to-peer network. Firstly, for multiple-version cache consideration in the transcoding-enable proxy, we focus on reducing the required server bandwidth and startup delay by caching the optimal versions of the video. A generalized video object profit function is derived from the extended weighted transcoding graph to calculate the individual cache profit of certain version of a video object, and the aggregate profit from caching multiple versions of the same video object. This proposed function takes into account the popularity of certain version of a video object, the transcoding delay among versions and the average access duration of each version. Based on the profit function, cache replacement algorithms are proposed to reduce the startup delay and network traffic by efficiently caching video objects with maximum profits. Next, a set of proxy-assisted transmission schemes are proposed to reduce the transmission cost for layered video streaming by integrating the proxy caching with reactive transmission schemes, peer-to-peer mesh networks and multicast capability. These proposed transmission schemes make multiple requests to be serviced by the single transmission and thus to significantly reduce the total required transmission cost. The optimal proxy prefix cache allocation is also calculated for each transmission scheme to identify the cache layers and cache length of each video to minimize the aggregate transmission cost. The process considers the fact that reduction in transmission cost by caching X layers of a video is not only from requests on X layers, but also from requests on less than X layers. Finally, we proposed a network coding equivalent content distribution (NCECD) scheme to decrease server stress, startup delay and jumping latency to support random access operations which are desirable for peer-to-peer on-demand video streaming. The random access operations are difficult to be efficiently supported, due to the asynchronous interactive behaviors of users and the dynamic nature of peers. In NCECD, videos are divided into segments which are then further divided into blocks. These blocks are then encoded into independent encoded blocks that are distributed to the local storage of different peers. With NCECD, a new client only needs to connect to a sufficient number of parent peers in order to view the whole video and rarely needs to find new parents when performing random access operations. Whereas most existing methods must search for parent peers containing interested segments, NCECD uses the properties of network coding to cache equivalent content on most peers, so that searches are rarely needed. The analysis of system parameters is given to achieve reasonable block loss rates for peer-to-peer interactive video-on-demand streaming. Experimental results demonstrate that these proposed schemes can lead to significant transmission cost saving, high delay saving ratio, high bandwidth saving ratio, low startup and jumping searching delays, connecting to a new parent peer delay and less server resources. Hence, these proposed schemes can further be integrated and utilized to build an efficient video streaming platform for providing high-performance and high-quality IPTV services to a diversity of clients.

This thesis presents an intelligent early packet discard (I-EPD) for real-time video streaming over a multi-hop ad hoc network. In a multi-hop ad hoc network, the quality of transferring real-time video streams could be seriously degraded, since every intermediate node (IN) functionally like forwarding device does not possess large buffer and sufficient bandwidth. Even worse, a selected forwarding node could leave or power off unexpectedly which breaks the route to destination. Thus, a video packet temporarily buffered in intermediate nodes may exceed its time constraint when either a congested or failed link occurs; a stale video packet is useless even if it can reach destination after network traffic becomes smooth or failed route is reconfigured. In the proposed I-EPD, an IN can intelligently determine whether a buffered video packet should be discarded based on an estimated time constraint which is calculated from the RTP timestamps and the round trip time (RTT) measured by RTCP. For the purpose of validation, we implement the I-EPD scheme on a Linux-based embedded system. We compare the quality of video streams under different bit rates and different route repair time. In addition, we use PSNR to validate the quality of pictures from the aspect of application layer. The experimental results demonstrate that with I-EPD buffer utilization on IN can be more effectively used and unnecessary bandwidth wastage can be avoided.

Peer-to-Peer (P2P) technology is a kind of decentralized framework which is applied to file sharing and video streaming in Internet predominantly. However, there are some problems, like free-riding, fairness, unexpected fluctuation, in P2P systems. Especially, without free riding controlling, many users stop contributing data and only want to receive data. To avoid these problems, a credit-based incentive mechanism using sneak, public-key cryptography and multi-source technique is proposed to make peers in system have no choice but to share data. In the proposed incentive mechanism, the concept of credit is employed. If a peer can contribute more bandwidth for sharing stream, it is arranged closer to video server, to get more credit in return, have less packet loss risk and receive better stream quality. Free-rider is a peer who denies contributing or contributes less resource than it registers to the system. In the proposed mechanism, free-riders are judged by the credits of each peer. In order to prevent illegal behaviors, the public-key cryptography is used for differentiating real and fake credit, also used for tracing credits. Experimental results show that when 90% peers are zero uploading peers, using the proposed mechanism can reduce 85% of zero uploading peers. When 90% peers are cheating peers, adopting the proposed mechanism can reduce 94% of cheating peers. System utility is usually about 100%. Maximum number of hops to video server is about five to six which is much smaller than without using the proposed mechanism, about 40 hops. The loads in managing system topology and delivering sub-streams are reduced. Besides, the resource of the system is used effectively.

Video traffic is growing rapidly in wireless networks. Different from ordinary data traffic, video streams have higher data rates and tighter delay constraints. The ever-varying throughput of wireless links, however, cannot support continuous video playback if the video data rate is kept at a high level. To this end, adaptive video transmission techniques are employed to reduce the risk of playback interruptions by dynamically matching the video data rate to the varying channel throughput. In this dissertation, I develop new models to capture viewers' quality of experience (QoE) and design adaptive transmission algorithms to optimize the QoE. The contributions of this dissertation are threefold. First, I develop a new model for the viewers' QoE in rate-switching systems in which the video source rate is adapted every several seconds. The model is developed to predict an important aspect of QoE, the time-varying subjective quality (TVSQ), i.e., the up-to-the-moment subjective quality of a video as it is played. I first build a video database of rate-switching videos and measure TVSQs via a subjective study. Then, I parameterize and validate the TVSQ model using the measured TVSQs. Finally, based on the TVSQ model, I design an adaptive rate-switching algorithm that optimizes the time-averaged TVSQs of wireless video users. Second, I propose an adaptive video transmission algorithm to optimize the Overall Quality (OQ) of rate-switching videos, i.e., the viewers' judgement on the quality of the whole video. Through the subjective study, I find that the OQ is strongly correlated with the empirical cumulative distribution function (eCDF) of the video quality perceived by viewers. Based on this observation, I develop an adaptive video transmission algorithm that maximizes the number of video users who satisfy given constraints on the eCDF of perceived video qualities. Third, I propose an adaptive transmission algorithm for scalable videos. Different from the rate-switching systems, scalable videos support rate adaptation for each video frame. The proposed adaptive transmission algorithm maximizes the time-averaged video quality while maintaining continuous video playback. When the channel throughput is high, the algorithm increases the video data rate to improve video quality. Otherwise, the algorithm decreases the video data rate to buffer more videos and to reduce the risk of playback interruption. Simulation results show that the performance of the proposed algorithm is close to a performance upper bound. / text

Pour avoir une idée sur la qualité du réseau, la majorité des acteurs concernés (opérateurs réseau, fournisseurs de service) se basent sur la Qualité de Service (Quality of Service). Cette mesure a montré des limites et beaucoup d'efforts ont été déployés pour mettre en place une nouvelle métrique qui reflète, de façon plus précise, la qualité du service offert. Cette mesure s'appelle la qualité d'expérience (Quality of Experience). La qualité d'expérience reflète la satisfaction de l'utilisateur par rapport au service qu'il utilise. Aujourd'hui, évaluer la qualité d'expérience est devenu primordiale pour les fournisseurs de services et les fournisseurs de contenus. Cette nécessité nous a poussés à innover et concevoir des nouvelles méthodes pour estimer la QoE. Dans cette thèse, nous travaillons sur l'estimation de la QoE (1) dans le cas des communications Voix sur IP et (2) dans le cas des services de diffusion Vidéo sur IP. Nous étudions les performances et la qualité des codecs iLBC, Speex et Silk pour la VoIP et les codecs MPEG-2 et H.264/SVC pour la vidéo sur IP. Nous étudions l'impact que peut avoir la majorité des paramètres réseaux, des paramètres sources (au niveau du codage) et destinations (au niveau du décodage) sur la qualité finale. Afin de mettre en place des outils précis d'estimation de la QoE en temps réel, nous nous basons sur la méthodologie Pseudo-Subjective Quality Assessment. La méthodologie PSQA est basée sur un modèle mathématique appelé les réseaux de neurones artificiels. En plus des réseaux de neurones, nous utilisons la régression polynomiale pour l'estimation de la QoE dans le cas de la VoIP.